September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Broken silence with webrtc
by Gary Foreman 01 Sep '15

01 Sep '15

I have that running. Unfortunately info reported in the console output appears to be the same whether the call has audio or not. The only message I've seen that has given a clue was "no audio stun for a long time". The setup is purely on a LAN so I'm not sure why stun is causing an issue. On Tue, Sep 1, 2015 at 4:25 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Ken Rice) > > > ---------- Forwarded message ---------- > From: Ken Rice <krice(a)freeswitch.org> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 10:25:10 -0500 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > > Just enable verto debugging in verto.conf.xml in your configs… it’ll print > it right to the screen > > > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Gary Foreman > *Sent:* Tuesday, September 1, 2015 10:13 AM > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Broken silence with webrtc > > > > Ok so the issue has been superseded by intermittent one-way / no audio. > I'm getting it very intermittently (1 in every 30 calls or so) but I'm > struggling to debug it as the traffic is encrypted and wireshark doesn't > see it as rtp stream. > > > > Where is the best place to start debugging verto? I was previously using > sip.js without any audio issues so it seems to be verto specific. > > > > On Tue, Sep 1, 2015 at 12:04 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Stanislav Sinyagin) > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 13:03:25 +0200 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > > is it running on a virtual machine? > > I found a strange effect that I could only reproduce in a VM, and never on > physical hardware: > https://freeswitch.org/jira/browse/FS-7805 > > under certain load, an originate command triggers a continuous distortion > in another, running and unrelated, channel. > > It seems to be triggered by insufficient CPU resource at the moment of the > origination. > > > > > > > > On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > > I've found that it occurs after any bridge, its not specific to the > originate command. > > > > Would you require a wireshark trace or the output of the freeswitch > console? > > > > The scenario below reproduces the issue ... > > > > Test extension > > > > <extension name="Inbound_Routing"> > > <condition field="destination_number" expression="^2003$"> > > <action application="export" data="dialed_extension=$1"/> > > <action application="export" data="transfer_ringback=$${uk-ring}"/> > > <action application="set" data="RECORD_TITLE=Title goes here"/> > > <action application="set" > data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> > > <action application="export" data="send_silence_when_idle=false"/> > > <action application="export" data="bridge_generate_comfort_noise=false"/> > > <action application="export" data="rtp_enable_vad_in=false"/> > > <action application="export" data="rtp_enable_vad_out=false"/> > > <action application="export" data="fire_talk_events=true"/> > > <action application="export" data="fire_not_talk_events=true"/> > > <action application="export" data="recording_id=${uuid}"/> > > <action application="set" data="media_bug_answer_req=true"/> > > <action application="answer" /> > > <action application="playback" > data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> > > </condition> > > </extension> > > > > I originate a call from a polycom handset using g722 to the extension > above. > > I originate a call using the verto client to the extension above. > > > > I get the uuids of the channels using show channels and use uuid_bridge > [uuid1] [uuid2] to merge the channels. > > > > On Tue, Sep 1, 2015 at 11:30 AM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. How send in To header anonymous and in P-Asserted-Identity > caller-id-number (Alex Polischuk) > 2. Loop play while wait DTMF digit (Dmitry Mordovin) > 3. Broken silence with webrtc (Gary Foreman) > 4. Re: Broken silence with webrtc (Brian West) > 5. SIP profile not loading (Chris Young) > 6. Playing with conditions (Dmitry Mordovin) > 7. Re: Playing with conditions (Avi Marcus) > > > ---------- Forwarded message ---------- > From: Alex Polischuk <alxpol(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 17:02:25 +0300 > Subject: [Freeswitch-users] How send in To header anonymous and in > P-Asserted-Identity caller-id-number > > Hi all, > > > > How I can define different users and domains in To and P-Asserted-Identity > headers? > > > > Thanks, > > Alex > > > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 18:24:08 +0400 > Subject: [Freeswitch-users] Loop play while wait DTMF digit > > Hello > > This example play conf-pin.wav and wait DTMF. > > > <extension name="play_and_get_digits example"> > > <condition field="destination_number" expression="^(1888)$"> > > <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> > > <action application="log" data="CRIT ${foobar}"/> > > </condition> > > </extension> > > > Is it possible to play WAV file in infinity loop and wait user DTMF? > > And how can I check DTMF input after user entered DTMF? > > In dialpeer, like. > If ${DTMF} = 1 then bridge to XXX > If ${DTMF} = 2 then play file and finish session > > > > Thank you. > Dmitry > > > > ---------- Forwarded message ---------- > From: Gary Foreman <gaz.foreman(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 31 Aug 2015 20:36:59 +0100 > Subject: [Freeswitch-users] Broken silence with webrtc > > Hi all, > > > > Hoping someone can point me in the right direction because after several > hours I'm out of ideas. > > > > I'm having an issue where the 2nd leg of a call that is bridged to a verto > endpoint has broken silence. When there is no sound on the line the audio > goes completely silent (all background noise is dropped) but around every > third of a second it repeatedly cuts back in for a fraction, then goes > completely silent again. This only happens when the call is created using > the originate command. > > > > Can anyone give me an idea of where to look next? I have a wireshark trace > that during playback shows the audio cutting out periodically on the 2nd > leg during periods of silence. The 1st leg is using webrtc encryption and I > cant decode the stream. > > > > Thanks in advance! > > > > > > > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 14:40:17 -0500 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > > Please provide logs and samples of how you originate this, sounds like > Voice Activity Detection possibly. > > > > On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > > Hi all, > > > > Hoping someone can point me in the right direction because after several > hours I'm out of ideas. > > > > I'm having an issue where the 2nd leg of a call that is bridged to a verto > endpoint has broken silence. When there is no sound on the line the audio > goes completely silent (all background noise is dropped) but around every > third of a second it repeatedly cuts back in for a fraction, then goes > completely silent again. This only happens when the call is created using > the originate command. > > > > Can anyone give me an idea of where to look next? I have a wireshark trace > that during playback shows the audio cutting out periodically on the 2nd > leg during periods of silence. The 1st leg is using webrtc encryption and I > cant decode the stream. > > > > Thanks in advance! > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian(a)freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > ---------- Forwarded message ---------- > From: Chris Young <Chris.Young(a)enghouse.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 07:31:00 +0000 > Subject: [Freeswitch-users] SIP profile not loading > > Hi all, > > > > Recently, we've begun experiencing a strange problem whereby the first SIP > profile to be loaded gets 'stuck' and never actually completes its > initialisation. This always seems to affect the first profile only, so if I > have profiles named (for example): > > > > dummy.xml > > external.xml > > internal.xml > > > > then 'dummy' would fail to load but 'external' and 'internal' would be > fine. No error messages are output to the logs but preliminary > investigation suggests that the profile thread is becoming blocked for some > reason. The specified IP address is valid and available and there are no > other processes using the requested port. FreeSWITCH comes up successfully > but 'sofia status' shows only 'external' and 'internal'. At this point, I > can use 'sofia profile dummy start' and the profile loads correctly so it > appears to be valid. > > > > Has anybody else seen this kind of behaviour or know what could be causing > it? > > > > Many thanks, > > Chris > > > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 01 Sep 2015 12:56:08 +0400 > Subject: [Freeswitch-users] Playing with conditions > Hello > > <section name="dialplan" description="Local call"> > <context name="public"> > <extension name="local_1"> > > <condition field="destination_number" expression="^(\d{4})$"> > <action application="answer"/> > > <action application="play_and_get_digits" data="1 1 3 7000 # > $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav > foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > > <condition field="${foobar}" expression="1$"> > <action application="log" data="CRIT > --------------1------------------"/> > <anti-action application="log" data="CRIT --------------not > 1------------------"/> > </condition> > > <action application="log" data="CRIT ---- last ${foobar}"/> > </condition> > > </extension> > </context> > </section> > > Why inner condition not works? > > > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 10:30:00 +0000 > Subject: Re: [Freeswitch-users] Playing with conditions > > Short answer: Each extension only has 1 set of conditions. > > The condition evaluating foobar is run *before* it gets set. > > After the play_and_get_digits you should transfer/execute_extension to a > new extension that will evaluate ${foobar} > > > -Avi Marcus > BestFone > > > > On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> > wrote: > > Hello > > <section name="dialplan" description="Local call"> > <context name="public"> > <extension name="local_1"> > > <condition field="destination_number" expression="^(\d{4})$"> > <action application="answer"/> > > <action application="play_and_get_digits" data="1 1 3 7000 # > $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav > /invalid.wav foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > > <condition field="${foobar}" expression="1$"> > <action application="log" data="CRIT > --------------1------------------"/> > <anti-action application="log" data="CRIT --------------not > 1------------------"/> > </condition> > > <action application="log" data="CRIT ---- last ${foobar}"/> > </condition> > > </extension> > </context> > </section> > > Why inner condition not works? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 0

Re: [Freeswitch-users] Broken silence with webrtc
by Gary Foreman 01 Sep '15

01 Sep '15

Ok so the issue has been superseded by intermittent one-way / no audio. I'm getting it very intermittently (1 in every 30 calls or so) but I'm struggling to debug it as the traffic is encrypted and wireshark doesn't see it as rtp stream. Where is the best place to start debugging verto? I was previously using sip.js without any audio issues so it seems to be verto specific. On Tue, Sep 1, 2015 at 12:04 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Stanislav Sinyagin) > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 13:03:25 +0200 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > is it running on a virtual machine? > > I found a strange effect that I could only reproduce in a VM, and never on > physical hardware: > https://freeswitch.org/jira/browse/FS-7805 > under certain load, an originate command triggers a continuous distortion > in another, running and unrelated, channel. > > It seems to be triggered by insufficient CPU resource at the moment of the > origination. > > > > > > > On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > >> I've found that it occurs after any bridge, its not specific to the >> originate command. >> >> Would you require a wireshark trace or the output of the freeswitch >> console? >> >> The scenario below reproduces the issue ... >> >> Test extension >> >> <extension name="Inbound_Routing"> >> <condition field="destination_number" expression="^2003$"> >> <action application="export" data="dialed_extension=$1"/> >> <action application="export" data="transfer_ringback=$${uk-ring}"/> >> <action application="set" data="RECORD_TITLE=Title goes here"/> >> <action application="set" >> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> >> <action application="export" data="send_silence_when_idle=false"/> >> <action application="export" data="bridge_generate_comfort_noise=false"/> >> <action application="export" data="rtp_enable_vad_in=false"/> >> <action application="export" data="rtp_enable_vad_out=false"/> >> <action application="export" data="fire_talk_events=true"/> >> <action application="export" data="fire_not_talk_events=true"/> >> <action application="export" data="recording_id=${uuid}"/> >> <action application="set" data="media_bug_answer_req=true"/> >> <action application="answer" /> >> <action application="playback" >> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> >> </condition> >> </extension> >> >> I originate a call from a polycom handset using g722 to the extension >> above. >> I originate a call using the verto client to the extension above. >> >> I get the uuids of the channels using show channels and use uuid_bridge >> [uuid1] [uuid2] to merge the channels. >> >> On Tue, Sep 1, 2015 at 11:30 AM, < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users(a)lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request(a)lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner(a)lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. How send in To header anonymous and in P-Asserted-Identity >>> caller-id-number (Alex Polischuk) >>> 2. Loop play while wait DTMF digit (Dmitry Mordovin) >>> 3. Broken silence with webrtc (Gary Foreman) >>> 4. Re: Broken silence with webrtc (Brian West) >>> 5. SIP profile not loading (Chris Young) >>> 6. Playing with conditions (Dmitry Mordovin) >>> 7. Re: Playing with conditions (Avi Marcus) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Alex Polischuk <alxpol(a)gmail.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 17:02:25 +0300 >>> Subject: [Freeswitch-users] How send in To header anonymous and in >>> P-Asserted-Identity caller-id-number >>> Hi all, >>> >>> How I can define different users and domains in To and P-Asserted-Identity >>> headers? >>> >>> Thanks, >>> Alex >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 18:24:08 +0400 >>> Subject: [Freeswitch-users] Loop play while wait DTMF digit >>> Hello >>> >>> This example play conf-pin.wav and wait DTMF. >>> >>> <extension name="play_and_get_digits example"> >>> <condition field="destination_number" expression="^(1888)$"> >>> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> </condition> >>> </extension> >>> >>> >>> Is it possible to play WAV file in infinity loop and wait user DTMF? >>> >>> And how can I check DTMF input after user entered DTMF? >>> >>> In dialpeer, like. >>> If ${DTMF} = 1 then bridge to XXX >>> If ${DTMF} = 2 then play file and finish session >>> >>> >>> >>> Thank you. >>> Dmitry >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gary Foreman <gaz.foreman(a)gmail.com> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Cc: >>> Date: Mon, 31 Aug 2015 20:36:59 +0100 >>> Subject: [Freeswitch-users] Broken silence with webrtc >>> Hi all, >>> >>> Hoping someone can point me in the right direction because after several >>> hours I'm out of ideas. >>> >>> I'm having an issue where the 2nd leg of a call that is bridged to a >>> verto endpoint has broken silence. When there is no sound on the line the >>> audio goes completely silent (all background noise is dropped) but around >>> every third of a second it repeatedly cuts back in for a fraction, then >>> goes completely silent again. This only happens when the call is created >>> using the originate command. >>> >>> Can anyone give me an idea of where to look next? I have a wireshark >>> trace that during playback shows the audio cutting out periodically on the >>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>> and I cant decode the stream. >>> >>> Thanks in advance! >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Brian West <brian(a)freeswitch.org> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 14:40:17 -0500 >>> Subject: Re: [Freeswitch-users] Broken silence with webrtc >>> Please provide logs and samples of how you originate this, sounds like >>> Voice Activity Detection possibly. >>> >>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> >>> wrote: >>> >>>> Hi all, >>>> >>>> Hoping someone can point me in the right direction because after >>>> several hours I'm out of ideas. >>>> >>>> I'm having an issue where the 2nd leg of a call that is bridged to a >>>> verto endpoint has broken silence. When there is no sound on the line the >>>> audio goes completely silent (all background noise is dropped) but around >>>> every third of a second it repeatedly cuts back in for a fraction, then >>>> goes completely silent again. This only happens when the call is created >>>> using the originate command. >>>> >>>> Can anyone give me an idea of where to look next? I have a wireshark >>>> trace that during playback shows the audio cutting out periodically on the >>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>>> and I cant decode the stream. >>>> >>>> Thanks in advance! >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian(a)freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: >>> /r/freeswitch <https://www.reddit.com/r/freeswitch> >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> >>> ---------- Forwarded message ---------- >>> From: Chris Young <Chris.Young(a)enghouse.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 07:31:00 +0000 >>> Subject: [Freeswitch-users] SIP profile not loading >>> >>> Hi all, >>> >>> >>> >>> Recently, we've begun experiencing a strange problem whereby the first >>> SIP profile to be loaded gets 'stuck' and never actually completes its >>> initialisation. This always seems to affect the first profile only, so if I >>> have profiles named (for example): >>> >>> >>> >>> dummy.xml >>> >>> external.xml >>> >>> internal.xml >>> >>> >>> >>> then 'dummy' would fail to load but 'external' and 'internal' would be >>> fine. No error messages are output to the logs but preliminary >>> investigation suggests that the profile thread is becoming blocked for some >>> reason. The specified IP address is valid and available and there are no >>> other processes using the requested port. FreeSWITCH comes up successfully >>> but 'sofia status' shows only 'external' and 'internal'. At this point, I >>> can use 'sofia profile dummy start' and the profile loads correctly so it >>> appears to be valid. >>> >>> >>> >>> Has anybody else seen this kind of behaviour or know what could be >>> causing it? >>> >>> >>> >>> Many thanks, >>> >>> Chris >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 01 Sep 2015 12:56:08 +0400 >>> Subject: [Freeswitch-users] Playing with conditions >>> Hello >>> >>> <section name="dialplan" description="Local call"> >>> <context name="public"> >>> <extension name="local_1"> >>> >>> <condition field="destination_number" expression="^(\d{4})$"> >>> <action application="answer"/> >>> >>> <action application="play_and_get_digits" data="1 1 3 7000 # >>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav >>> foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> >>> <condition field="${foobar}" expression="1$"> >>> <action application="log" data="CRIT >>> --------------1------------------"/> >>> <anti-action application="log" data="CRIT --------------not >>> 1------------------"/> >>> </condition> >>> >>> <action application="log" data="CRIT ---- last ${foobar}"/> >>> </condition> >>> >>> </extension> >>> </context> >>> </section> >>> >>> Why inner condition not works? >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus <avi(a)avimarcus.net> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 10:30:00 +0000 >>> Subject: Re: [Freeswitch-users] Playing with conditions >>> Short answer: Each extension only has 1 set of conditions. >>> The condition evaluating foobar is run *before* it gets set. >>> After the play_and_get_digits you should transfer/execute_extension to >>> a new extension that will evaluate ${foobar} >>> >>> -Avi Marcus >>> BestFone >>> >>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> >>> wrote: >>> >>>> Hello >>>> >>>> <section name="dialplan" description="Local call"> >>>> <context name="public"> >>>> <extension name="local_1"> >>>> >>>> <condition field="destination_number" expression="^(\d{4})$"> >>>> <action application="answer"/> >>>> >>>> <action application="play_and_get_digits" data="1 1 3 7000 # >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >>>> /invalid.wav foobar \d+"/> >>>> <action application="log" data="CRIT ${foobar}"/> >>>> >>>> <condition field="${foobar}" expression="1$"> >>>> <action application="log" data="CRIT >>>> --------------1------------------"/> >>>> <anti-action application="log" data="CRIT --------------not >>>> 1------------------"/> >>>> </condition> >>>> >>>> <action application="log" data="CRIT ---- last ${foobar}"/> >>>> </condition> >>>> >>>> </extension> >>>> </context> >>>> </section> >>>> >>>> Why inner condition not works? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

3 2

Playing with conditions
by Dmitry Mordovin 01 Sep '15

01 Sep '15

Hello <section name="dialplan" description="Local call"> <context name="public"> <extension name="local_1"> <condition field="destination_number" expression="^(\d{4})$"> <action application="answer"/> <action application="play_and_get_digits" data="1 1 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/> <condition field="${foobar}" expression="1$"> <action application="log" data="CRIT --------------1------------------"/> <anti-action application="log" data="CRIT --------------not 1------------------"/> </condition> <action application="log" data="CRIT ---- last ${foobar}"/> </condition> </extension> </context> </section> Why inner condition not works?

3 3

SIP profile not loading
by Chris Young 01 Sep '15

01 Sep '15

Hi all, Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named (for example): dummy.xml external.xml internal.xml then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly so it appears to be valid. Has anybody else seen this kind of behaviour or know what could be causing it? Many thanks, Chris

3 3

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 3
by Gary Foreman 01 Sep '15

01 Sep '15

I'm getting the issue on a physical server with no load unfortunately. I'm trying to build the latest FS to test but its proving difficult on CentOS 6, looking like a Debian VM is the next step. On Tue, Sep 1, 2015 at 12:04 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Stanislav Sinyagin) > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 13:03:25 +0200 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > is it running on a virtual machine? > > I found a strange effect that I could only reproduce in a VM, and never on > physical hardware: > https://freeswitch.org/jira/browse/FS-7805 > under certain load, an originate command triggers a continuous distortion > in another, running and unrelated, channel. > > It seems to be triggered by insufficient CPU resource at the moment of the > origination. > > > > > > > On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > >> I've found that it occurs after any bridge, its not specific to the >> originate command. >> >> Would you require a wireshark trace or the output of the freeswitch >> console? >> >> The scenario below reproduces the issue ... >> >> Test extension >> >> <extension name="Inbound_Routing"> >> <condition field="destination_number" expression="^2003$"> >> <action application="export" data="dialed_extension=$1"/> >> <action application="export" data="transfer_ringback=$${uk-ring}"/> >> <action application="set" data="RECORD_TITLE=Title goes here"/> >> <action application="set" >> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> >> <action application="export" data="send_silence_when_idle=false"/> >> <action application="export" data="bridge_generate_comfort_noise=false"/> >> <action application="export" data="rtp_enable_vad_in=false"/> >> <action application="export" data="rtp_enable_vad_out=false"/> >> <action application="export" data="fire_talk_events=true"/> >> <action application="export" data="fire_not_talk_events=true"/> >> <action application="export" data="recording_id=${uuid}"/> >> <action application="set" data="media_bug_answer_req=true"/> >> <action application="answer" /> >> <action application="playback" >> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> >> </condition> >> </extension> >> >> I originate a call from a polycom handset using g722 to the extension >> above. >> I originate a call using the verto client to the extension above. >> >> I get the uuids of the channels using show channels and use uuid_bridge >> [uuid1] [uuid2] to merge the channels. >> >> On Tue, Sep 1, 2015 at 11:30 AM, < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users(a)lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request(a)lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner(a)lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. How send in To header anonymous and in P-Asserted-Identity >>> caller-id-number (Alex Polischuk) >>> 2. Loop play while wait DTMF digit (Dmitry Mordovin) >>> 3. Broken silence with webrtc (Gary Foreman) >>> 4. Re: Broken silence with webrtc (Brian West) >>> 5. SIP profile not loading (Chris Young) >>> 6. Playing with conditions (Dmitry Mordovin) >>> 7. Re: Playing with conditions (Avi Marcus) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Alex Polischuk <alxpol(a)gmail.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 17:02:25 +0300 >>> Subject: [Freeswitch-users] How send in To header anonymous and in >>> P-Asserted-Identity caller-id-number >>> Hi all, >>> >>> How I can define different users and domains in To and P-Asserted-Identity >>> headers? >>> >>> Thanks, >>> Alex >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 18:24:08 +0400 >>> Subject: [Freeswitch-users] Loop play while wait DTMF digit >>> Hello >>> >>> This example play conf-pin.wav and wait DTMF. >>> >>> <extension name="play_and_get_digits example"> >>> <condition field="destination_number" expression="^(1888)$"> >>> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> </condition> >>> </extension> >>> >>> >>> Is it possible to play WAV file in infinity loop and wait user DTMF? >>> >>> And how can I check DTMF input after user entered DTMF? >>> >>> In dialpeer, like. >>> If ${DTMF} = 1 then bridge to XXX >>> If ${DTMF} = 2 then play file and finish session >>> >>> >>> >>> Thank you. >>> Dmitry >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gary Foreman <gaz.foreman(a)gmail.com> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Cc: >>> Date: Mon, 31 Aug 2015 20:36:59 +0100 >>> Subject: [Freeswitch-users] Broken silence with webrtc >>> Hi all, >>> >>> Hoping someone can point me in the right direction because after several >>> hours I'm out of ideas. >>> >>> I'm having an issue where the 2nd leg of a call that is bridged to a >>> verto endpoint has broken silence. When there is no sound on the line the >>> audio goes completely silent (all background noise is dropped) but around >>> every third of a second it repeatedly cuts back in for a fraction, then >>> goes completely silent again. This only happens when the call is created >>> using the originate command. >>> >>> Can anyone give me an idea of where to look next? I have a wireshark >>> trace that during playback shows the audio cutting out periodically on the >>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>> and I cant decode the stream. >>> >>> Thanks in advance! >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Brian West <brian(a)freeswitch.org> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Mon, 31 Aug 2015 14:40:17 -0500 >>> Subject: Re: [Freeswitch-users] Broken silence with webrtc >>> Please provide logs and samples of how you originate this, sounds like >>> Voice Activity Detection possibly. >>> >>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> >>> wrote: >>> >>>> Hi all, >>>> >>>> Hoping someone can point me in the right direction because after >>>> several hours I'm out of ideas. >>>> >>>> I'm having an issue where the 2nd leg of a call that is bridged to a >>>> verto endpoint has broken silence. When there is no sound on the line the >>>> audio goes completely silent (all background noise is dropped) but around >>>> every third of a second it repeatedly cuts back in for a fraction, then >>>> goes completely silent again. This only happens when the call is created >>>> using the originate command. >>>> >>>> Can anyone give me an idea of where to look next? I have a wireshark >>>> trace that during playback shows the audio cutting out periodically on the >>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >>>> and I cant decode the stream. >>>> >>>> Thanks in advance! >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian(a)freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: >>> /r/freeswitch <https://www.reddit.com/r/freeswitch> >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> >>> ---------- Forwarded message ---------- >>> From: Chris Young <Chris.Young(a)enghouse.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 07:31:00 +0000 >>> Subject: [Freeswitch-users] SIP profile not loading >>> >>> Hi all, >>> >>> >>> >>> Recently, we've begun experiencing a strange problem whereby the first >>> SIP profile to be loaded gets 'stuck' and never actually completes its >>> initialisation. This always seems to affect the first profile only, so if I >>> have profiles named (for example): >>> >>> >>> >>> dummy.xml >>> >>> external.xml >>> >>> internal.xml >>> >>> >>> >>> then 'dummy' would fail to load but 'external' and 'internal' would be >>> fine. No error messages are output to the logs but preliminary >>> investigation suggests that the profile thread is becoming blocked for some >>> reason. The specified IP address is valid and available and there are no >>> other processes using the requested port. FreeSWITCH comes up successfully >>> but 'sofia status' shows only 'external' and 'internal'. At this point, I >>> can use 'sofia profile dummy start' and the profile loads correctly so it >>> appears to be valid. >>> >>> >>> >>> Has anybody else seen this kind of behaviour or know what could be >>> causing it? >>> >>> >>> >>> Many thanks, >>> >>> Chris >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >>> To: "freeswitch-users(a)lists.freeswitch.org" < >>> freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 01 Sep 2015 12:56:08 +0400 >>> Subject: [Freeswitch-users] Playing with conditions >>> Hello >>> >>> <section name="dialplan" description="Local call"> >>> <context name="public"> >>> <extension name="local_1"> >>> >>> <condition field="destination_number" expression="^(\d{4})$"> >>> <action application="answer"/> >>> >>> <action application="play_and_get_digits" data="1 1 3 7000 # >>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav >>> foobar \d+"/> >>> <action application="log" data="CRIT ${foobar}"/> >>> >>> <condition field="${foobar}" expression="1$"> >>> <action application="log" data="CRIT >>> --------------1------------------"/> >>> <anti-action application="log" data="CRIT --------------not >>> 1------------------"/> >>> </condition> >>> >>> <action application="log" data="CRIT ---- last ${foobar}"/> >>> </condition> >>> >>> </extension> >>> </context> >>> </section> >>> >>> Why inner condition not works? >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus <avi(a)avimarcus.net> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Tue, 1 Sep 2015 10:30:00 +0000 >>> Subject: Re: [Freeswitch-users] Playing with conditions >>> Short answer: Each extension only has 1 set of conditions. >>> The condition evaluating foobar is run *before* it gets set. >>> After the play_and_get_digits you should transfer/execute_extension to >>> a new extension that will evaluate ${foobar} >>> >>> -Avi Marcus >>> BestFone >>> >>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> >>> wrote: >>> >>>> Hello >>>> >>>> <section name="dialplan" description="Local call"> >>>> <context name="public"> >>>> <extension name="local_1"> >>>> >>>> <condition field="destination_number" expression="^(\d{4})$"> >>>> <action application="answer"/> >>>> >>>> <action application="play_and_get_digits" data="1 1 3 7000 # >>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >>>> /invalid.wav foobar \d+"/> >>>> <action application="log" data="CRIT ${foobar}"/> >>>> >>>> <condition field="${foobar}" expression="1$"> >>>> <action application="log" data="CRIT >>>> --------------1------------------"/> >>>> <anti-action application="log" data="CRIT --------------not >>>> 1------------------"/> >>>> </condition> >>>> >>>> <action application="log" data="CRIT ---- last ${foobar}"/> >>>> </condition> >>>> >>>> </extension> >>>> </context> >>>> </section> >>>> >>>> Why inner condition not works? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

2 1

Loop play while wait DTMF digit
by Dmitry Mordovin 01 Sep '15

01 Sep '15

Hello This example play conf-pin.wav and wait DTMF. <extension name="play_and_get_digits example"> <condition field="destination_number" expression="^(1888)$"> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/> </condition> </extension> Is it possible to play WAV file in infinity loop and wait user DTMF? And how can I check DTMF input after user entered DTMF? In dialpeer, like. If ${DTMF} = 1 then bridge to XXX If ${DTMF} = 2 then play file and finish session Thank you. Dmitry

2 1

Re: [Freeswitch-users] Broken silence with webrtc
by Gary Foreman 01 Sep '15

01 Sep '15

I've found that it occurs after any bridge, its not specific to the originate command. Would you require a wireshark trace or the output of the freeswitch console? The scenario below reproduces the issue ... Test extension <extension name="Inbound_Routing"> <condition field="destination_number" expression="^2003$"> <action application="export" data="dialed_extension=$1"/> <action application="export" data="transfer_ringback=$${uk-ring}"/> <action application="set" data="RECORD_TITLE=Title goes here"/> <action application="set" data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> <action application="export" data="send_silence_when_idle=false"/> <action application="export" data="bridge_generate_comfort_noise=false"/> <action application="export" data="rtp_enable_vad_in=false"/> <action application="export" data="rtp_enable_vad_out=false"/> <action application="export" data="fire_talk_events=true"/> <action application="export" data="fire_not_talk_events=true"/> <action application="export" data="recording_id=${uuid}"/> <action application="set" data="media_bug_answer_req=true"/> <action application="answer" /> <action application="playback" data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> </condition> </extension> I originate a call from a polycom handset using g722 to the extension above. I originate a call using the verto client to the extension above. I get the uuids of the channels using show channels and use uuid_bridge [uuid1] [uuid2] to merge the channels. On Tue, Sep 1, 2015 at 11:30 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. How send in To header anonymous and in P-Asserted-Identity > caller-id-number (Alex Polischuk) > 2. Loop play while wait DTMF digit (Dmitry Mordovin) > 3. Broken silence with webrtc (Gary Foreman) > 4. Re: Broken silence with webrtc (Brian West) > 5. SIP profile not loading (Chris Young) > 6. Playing with conditions (Dmitry Mordovin) > 7. Re: Playing with conditions (Avi Marcus) > > > ---------- Forwarded message ---------- > From: Alex Polischuk <alxpol(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 17:02:25 +0300 > Subject: [Freeswitch-users] How send in To header anonymous and in > P-Asserted-Identity caller-id-number > Hi all, > > How I can define different users and domains in To and P-Asserted-Identity > headers? > > Thanks, > Alex > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 18:24:08 +0400 > Subject: [Freeswitch-users] Loop play while wait DTMF digit > Hello > > This example play conf-pin.wav and wait DTMF. > > <extension name="play_and_get_digits example"> > <condition field="destination_number" expression="^(1888)$"> > <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > </condition> > </extension> > > > Is it possible to play WAV file in infinity loop and wait user DTMF? > > And how can I check DTMF input after user entered DTMF? > > In dialpeer, like. > If ${DTMF} = 1 then bridge to XXX > If ${DTMF} = 2 then play file and finish session > > > > Thank you. > Dmitry > > > > ---------- Forwarded message ---------- > From: Gary Foreman <gaz.foreman(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 31 Aug 2015 20:36:59 +0100 > Subject: [Freeswitch-users] Broken silence with webrtc > Hi all, > > Hoping someone can point me in the right direction because after several > hours I'm out of ideas. > > I'm having an issue where the 2nd leg of a call that is bridged to a verto > endpoint has broken silence. When there is no sound on the line the audio > goes completely silent (all background noise is dropped) but around every > third of a second it repeatedly cuts back in for a fraction, then goes > completely silent again. This only happens when the call is created using > the originate command. > > Can anyone give me an idea of where to look next? I have a wireshark trace > that during playback shows the audio cutting out periodically on the 2nd > leg during periods of silence. The 1st leg is using webrtc encryption and I > cant decode the stream. > > Thanks in advance! > > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Aug 2015 14:40:17 -0500 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > Please provide logs and samples of how you originate this, sounds like > Voice Activity Detection possibly. > > On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> > wrote: > >> Hi all, >> >> Hoping someone can point me in the right direction because after several >> hours I'm out of ideas. >> >> I'm having an issue where the 2nd leg of a call that is bridged to a >> verto endpoint has broken silence. When there is no sound on the line the >> audio goes completely silent (all background noise is dropped) but around >> every third of a second it repeatedly cuts back in for a fraction, then >> goes completely silent again. This only happens when the call is created >> using the originate command. >> >> Can anyone give me an idea of where to look next? I have a wireshark >> trace that during playback shows the audio cutting out periodically on the >> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >> and I cant decode the stream. >> >> Thanks in advance! >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian(a)freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > ---------- Forwarded message ---------- > From: Chris Young <Chris.Young(a)enghouse.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 07:31:00 +0000 > Subject: [Freeswitch-users] SIP profile not loading > > Hi all, > > > > Recently, we've begun experiencing a strange problem whereby the first SIP > profile to be loaded gets 'stuck' and never actually completes its > initialisation. This always seems to affect the first profile only, so if I > have profiles named (for example): > > > > dummy.xml > > external.xml > > internal.xml > > > > then 'dummy' would fail to load but 'external' and 'internal' would be > fine. No error messages are output to the logs but preliminary > investigation suggests that the profile thread is becoming blocked for some > reason. The specified IP address is valid and available and there are no > other processes using the requested port. FreeSWITCH comes up successfully > but 'sofia status' shows only 'external' and 'internal'. At this point, I > can use 'sofia profile dummy start' and the profile loads correctly so it > appears to be valid. > > > > Has anybody else seen this kind of behaviour or know what could be causing > it? > > > > Many thanks, > > Chris > > > > > ---------- Forwarded message ---------- > From: Dmitry Mordovin <d.mordovin(a)dwide.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 01 Sep 2015 12:56:08 +0400 > Subject: [Freeswitch-users] Playing with conditions > Hello > > <section name="dialplan" description="Local call"> > <context name="public"> > <extension name="local_1"> > > <condition field="destination_number" expression="^(\d{4})$"> > <action application="answer"/> > > <action application="play_and_get_digits" data="1 1 3 7000 # > $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav > foobar \d+"/> > <action application="log" data="CRIT ${foobar}"/> > > <condition field="${foobar}" expression="1$"> > <action application="log" data="CRIT > --------------1------------------"/> > <anti-action application="log" data="CRIT --------------not > 1------------------"/> > </condition> > > <action application="log" data="CRIT ---- last ${foobar}"/> > </condition> > > </extension> > </context> > </section> > > Why inner condition not works? > > > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 10:30:00 +0000 > Subject: Re: [Freeswitch-users] Playing with conditions > Short answer: Each extension only has 1 set of conditions. > The condition evaluating foobar is run *before* it gets set. > After the play_and_get_digits you should transfer/execute_extension to a > new extension that will evaluate ${foobar} > > -Avi Marcus > BestFone > > On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> > wrote: > >> Hello >> >> <section name="dialplan" description="Local call"> >> <context name="public"> >> <extension name="local_1"> >> >> <condition field="destination_number" expression="^(\d{4})$"> >> <action application="answer"/> >> >> <action application="play_and_get_digits" data="1 1 3 7000 # >> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >> /invalid.wav foobar \d+"/> >> <action application="log" data="CRIT ${foobar}"/> >> >> <condition field="${foobar}" expression="1$"> >> <action application="log" data="CRIT >> --------------1------------------"/> >> <anti-action application="log" data="CRIT --------------not >> 1------------------"/> >> </condition> >> >> <action application="log" data="CRIT ---- last ${foobar}"/> >> </condition> >> >> </extension> >> </context> >> </section> >> >> Why inner condition not works? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

2 1